Abstract: The evolution in mobile broadband technologies has allowed to increase the download rates in users considering the current services. The evaluation of technical parameters at the link level is of vital importance to validate the quality and veracity of the connection, thus avoiding large losses of data, time and productivity. Some of these failures may occur between the eNodeB (Evolved Node B) and the user equipment (UE), so the link between the end device and the base station can be observed. LTE (Long Term Evolution) is considered one of the IP-oriented mobile broadband technologies that work stably for data and VoIP (Voice Over IP) for those devices that have that feature. This research presents a technical analysis of the connection and channeling processes between UE and eNodeB with the TAC (Tracking Area Code) variables, and analysis of performance variables (Throughput, Signal to Interference and Noise Ratio (SINR)). Three measurement scenarios were proposed in the city of Bogotá using QualiPoc, where two operators were evaluated (Operator 1 and Operator 2). Once the data were obtained, an analysis of the variables was performed determining that the data obtained in transmission modes vary depending on the parameters BLER (Block Error Rate), performance and SNR (Signal-to-Noise Ratio). In the case of both operators, differences in transmission modes are detected and this is reflected in the quality of the signal. In addition, due to the fact that both operators work in different frequencies, it can be seen that Operator 1, despite having spectrum in Band 7 (2600 MHz), together with Operator 2, is reassigning to another frequency, a lower band, which is AWS (1700 MHz), but the difference in signal quality with respect to the establishment with data by the provider Operator 2 and the difference found in the transmission modes determined by the eNodeB in Operator 1 is remarkable.
Abstract: Steganalysis seeks to detect the presence of secret data embedded in cover objects, and there is an imminent demand to detect hidden messages in streaming media. This paper shows how a steganalysis algorithm based on Fast Fourier Transform (FFT) can be used to detect the existence of secret data embedded in streaming media. The proposed algorithm uses machine parameter characteristics and a network sniffer to determine whether the Internet traffic contains streaming channels. The detected streaming data is then transferred from the time domain to the frequency domain through FFT. The distributions of power spectra in the frequency domain between original VoIP streams and stego VoIP streams are compared in turn using t-test, achieving the p-value of 7.5686E-176 which is below the threshold. The results indicate that the proposed FFT-based steganalysis algorithm is effective in detecting the secret data embedded in VoIP streaming media.
Abstract: Nowadays, security threats in Voice over IP (VoIP) systems are an essential and latent concern for people in charge of security in a corporate network, because, every day, new Denial-of-Service (DoS) attacks are developed. These affect the business continuity of an organization, regarding confidentiality, availability, and integrity of services, causing frequent losses of both information and money. The purpose of this study is to establish the necessary measures to mitigate DoS threats, which affect the availability of VoIP systems, based on the Session Initiation Protocol (SIP). A Security Model called MS-DoS-SIP is proposed, which is based on two approaches. The first one analyzes the recommendations of international security standards. The second approach takes into account weaknesses and threats. The implementation of this model in a VoIP simulated system allowed to minimize the present vulnerabilities in 92% and increase the availability time of the VoIP service into an organization.
Abstract: Speech enhancement is a long standing problem with
numerous applications like teleconferencing, VoIP, hearing aids and
speech recognition. The motivation behind this research work is to
obtain a clean speech signal of higher quality by applying the optimal
noise cancellation technique. Real-time adaptive filtering algorithms
seem to be the best candidate among all categories of the speech
enhancement methods. In this paper, we propose a speech
enhancement method based on Recursive Least Squares (RLS)
adaptive filter of speech signals. Experiments were performed on
noisy data which was prepared by adding AWGN, Babble and Pink
noise to clean speech samples at -5dB, 0dB, 5dB and 10dB SNR
levels. We then compare the noise cancellation performance of
proposed RLS algorithm with existing NLMS algorithm in terms of
Mean Squared Error (MSE), Signal to Noise ratio (SNR) and SNR
Loss. Based on the performance evaluation, the proposed RLS
algorithm was found to be a better optimal noise cancellation
technique for speech signals.
Abstract: Scheduling is the process of dynamically allocating physical resources to User Equipment (UE) based on scheduling algorithms implemented at the LTE base station. Various algorithms have been proposed by network researchers as the implementation of scheduling algorithm which represents an open issue in Long Term Evolution (LTE) standard. This paper makes an attempt to study and compare the performance of PF, MLWDF and EXP/PF scheduling algorithms. The evaluation is considered for a single cell with interference scenario for different flows such as Best effort, Video and VoIP in a pedestrian and vehicular environment using the LTE-Sim network simulator. The comparative study is conducted in terms of system throughput, fairness index, delay, packet loss ratio (PLR) and total cell spectral efficiency.
Abstract: Voice over Internet Protocol (VoIP) products is an emerging technology that can contain forensically important information for a criminal activity. Without having the user name and passwords, this forensically important information can still be gathered by the investigators. Although there are a few VoIP forensic investigative applications available in the literature, most of them are particularly designed to collect evidence from the Skype product. Therefore, in order to assist law enforcement with collecting forensically important information from variety of Betamax VoIP tools, CVOIP-FRU framework is developed. CVOIP-FRU provides a data gathering solution that retrieves usernames, contact lists, as well as call and SMS logs from Betamax VoIP products. It is a scripting utility that searches for data within the registry, logs and the user roaming profiles in Windows and Mac OSX operating systems. Subsequently, it parses the output into readable text and html formats. One superior way of CVOIP-FRU compared to the other applications that due to intelligent data filtering capabilities and cross platform scripting back end of CVOIP-FRU, it is expandable to include other VoIP solutions as well. Overall, this paper reveals the exploratory analysis performed in order to find the key data paths and locations, the development stages of the framework, and the empirical testing and quality assurance of CVOIP-FRU.
Abstract: Voice Over IP (VoIP) is a technology that could pass
the voice traffic and data packet form over an IP network. Network
can be used for intranet or Internet. Phone calls using VoIP has
advantages in terms of cheaper cost of PSTN phone to more than
half, because the cost is calculated by the cost of the global nature of
the Internet. Session Initiation Protocol (SIP) is a signaling protocol
at the application layer which serves to establish, modify, and
terminate a multimedia session involving one or more users. This SIP
signaling has SIP message in text form that is used for session
management by the SIP components, such as User Agent, Registrar,
Redirect Server, and Proxy Server. To build a SIP communication is
required SIP Express Router (SER) to be able to receive SIP
messages, for handling the basic functions of SIP messages.
Problems occur when the NAT through which affects the voice
communication will be blocked starting from the sound that is not
sent or one side of the sound are sent (half duplex). How that could
be used to penetrate NAT is to use a given mediaproxy random RTP
port to penetrate NAT.
Abstract: Wireless LAN technologies have picked up
momentum in the recent years due to their ease of deployment, cost
and availability. The era of wireless LAN has also given rise to
unique applications like VOIP, IPTV and unified messaging.
However, these real-time applications are very sensitive to network
and handoff latencies. To successfully support these applications,
seamless roaming during the movement of mobile station has become
crucial. Nowadays, centralized architecture models support roaming
in WLANs. They have the ability to manage, control and
troubleshoot large scale WLAN deployments. This model is managed
by Control and Provision of Wireless Access Point protocol
(CAPWAP). This paper covers the CAPWAP architectural solution
along with its proposals that have emerged. Based on the literature
survey conducted in this paper, we found that the proposed
algorithms to reduce roaming latency in CAPWAP architecture do
not support seamless roaming. Additionally, they are not sufficient
during the initial period of the network. This paper also suggests
important design consideration for mobility support in future
centralized IEEE 802.11 networks.
Abstract: In this work, a characterization and modeling of
packet loss of a Voice over Internet Protocol (VoIP) communication
is developed. The distributions of the number of consecutive received
and lost packets (namely gap and burst) are modeled from the
transition probabilities of two-state and four-state model.
Measurements show that both models describe adequately the burst
distribution, but the decay of gap distribution for non-homogeneous
losses is better fit by the four-state model. The respective
probabilities of transition between states for each model were
estimated with a proposed algorithm from a set of monitored VoIP
calls in order to obtain representative minimum, maximum and
average values for both models.
Abstract: Today-s Wi Fi generation utilize the latest technology in their daily lives. Instructors at National University, the second largest non profit private institution of higher learning in California, are incorporating these new tools to modify their Online class formats to better accommodate these new skills in their distance education delivery modes. The University provides accelerated learning in a one-course per month format both Onsite and Online. Since there has been such a significant increase in Online classes over the past three years, and it is expected to grow even more over the over the next five years, Instructors cannot afford to maintain the status quo and not take advantage of these new options. It is at the discretion of the instructors which accessory they use and how comfortable and familiar they are with the technology. This paper explores the effects and summarizes students- comments of some of these new technological options which have been recently provided in order to make students- online learning experience more exciting and meaningful.
Abstract: Voice over Internet Protocol (VoIP) is a form of voice
communication that uses audio data to transmit voice signals to the
end user. VoIP is one of the most important technologies in the
World of communication. Around, 20 years of research on VoIP,
some problems of VoIP are still remaining. During the past decade
and with growing of wireless technologies, we have seen that many
papers turn their concentration from Wired-LAN to Wireless-LAN.
VoIP over Wireless LAN (WLAN) faces many challenges due to the
loose nature of wireless network. Issues like providing Quality of
Service (QoS) at a good level, dedicating capacity for calls and
having secure calls is more difficult rather than wired LAN.
Therefore VoIP over WLAN (VoWLAN) remains a challenging
research topic. In this paper we consolidate and address major
VoWLAN issues. This research is helpful for those researchers wants
to do research in Voice over IP technology over WLAN network.
Abstract: The VoIP networks as alternative method to traditional PSTN system has been implemented in a wide variety of structures
with multiple protocols, codecs, software and hardware–based
distributions. The use of cryptographic techniques let the users to have a secure communication, but the calculate throughput as well as the QoS parameters are affected according to the used algorithm. This
paper analyzes the VoIP throughput and the QoS parameters with
different commercial encryption methods. The measurement–based
approach uses lab scenarios to simulate LAN and WAN
environments. Security mechanisms such as TLS, SIAX2, SRTP,
IPSEC and ZRTP are analyzed with μ-LAW and GSM codecs.
Abstract: SIP (Session Initiation Protocol), using HTML based
call control messaging which is quite simple and efficient, is being
replaced for VoIP networks recently. As for authentication and
authorization purposes there are many approaches and considerations
for securing SIP to eliminate forgery on the integrity of SIP
messages. On the other hand Elliptic Curve Cryptography has
significant advantages like smaller key sizes, faster computations on
behalf of other Public Key Cryptography (PKC) systems that obtain
data transmission more secure and efficient. In this work a new
approach is proposed for secure SIP authentication by using a public
key exchange mechanism using ECC. Total execution times and
memory requirements of proposed scheme have been improved in
comparison with non-elliptic approaches by adopting elliptic-based
key exchange mechanism.
Abstract: Multiparty voice over IP (MVoIP) systems allows a group of people to freely communicate each other via the internet, which have many applications such as online gaming, teleconferencing, online stock trading etc. Peertalk is a peer to peer multiparty voice over IP system (MVoIP) which is more feasible than existing approaches such as p2p overlay multicast and coupled distributed processing. Since the stream mixing and distribution are done by the peers, it is vulnerable to major security threats like nodes misbehavior, eavesdropping, Sybil attacks, Denial of Service (DoS), call tampering, Man in the Middle attacks etc. To thwart the security threats, a security framework called PEERTS (PEEred Reputed Trustworthy System for peertalk) is implemented so that efficient and secure communication can be carried out between peers.
Abstract: In this paper we present a statistical analysis of Voice
over IP (VoIP) packet streams produced by the G.711 voice coder
with voice activity detection (VAD). During telephone conversation,
depending whether the interlocutor speaks (ON) or remains silent
(OFF), packets are produced or not by a voice coder. As index of
dispersion for both ON and OFF times distribution was greater than
one, we used hyperexponential distribution for approximation of
streams duration. For each stage of the hyperexponential distribution,
we tested goodness of our fits using graphical methods, we calculated
estimation errors, and performed Kolmogorov-Smirnov test.
Obtained results showed that the precise VoIP source model can be
based on the five-state Markov process.
Abstract: Nowadays, Multimedia Communication has been developed and improved rapidly in order to enable users to communicate between each other over the Internet. In general, the multimedia communication consists of audio and video communication. However, this paper focuses on audio streams. The audio translation between protocols is a very critical issue due to solving the communication problems between any two protocols, as well as it enables people around the world to talk with each other at anywhere and anytime even they use different protocols. In this paper, a proposed method for an audio translation module between two protocols has been presented. These two protocols are InterAsterisk eXchange Protocol (IAX) and Real Time Switching Control Protocol (RSW), which they are widely used to provide two ways audio transfer feature. The result of this work is to introduce possibility of interworking together.
Abstract: In this paper, a new encoding algorithm of spectral envelope based on NLMS in G.729.1 for VoIP is proposed. In the TDAC part of G.729.1, the spectral envelope and MDCT coefficients extracted in the weighted CELP coding error (lower-band) and the higher-band input signal are encoded. In order to reduce allocation bits for spectral envelope coding, a new quantization algorithm based on NLMS is proposed. Also, reduced bits are used to enhance sound quality. The performance of the proposed algorithm is evaluated by sound quality and bit reduction rates in clean and frame loss conditions.
Abstract: The voice signal in Voice over Internet protocol (VoIP) system is processed through the best effort policy based IP network, which leads to the network degradations including delay, packet loss jitter. The work in this paper presents the implementation of finite impulse response (FIR) filter for voice quality improvement in the VoIP system through distributed arithmetic (DA) algorithm. The VoIP simulations are conducted with AMR-NB 6.70 kbps and G.729a speech coders at different packet loss rates and the performance of the enhanced VoIP signal is evaluated using the perceptual evaluation of speech quality (PESQ) measurement for narrowband signal. The results show reduction in the computational complexity in the system and significant improvement in the quality of the VoIP voice signal.
Abstract: This paper is taken into consideration the problem of cryptanalysis of stream ciphers. There is some attempts need to improve the existing attacks on stream cipher and to make an attempt to distinguish the portions of cipher text obtained by the encryption of plain text in which some parts of the text are random and the rest are non-random. This paper presents a tutorial introduction to symmetric cryptography. The basic information theoretic and computational properties of classic and modern cryptographic systems are presented, followed by an examination of the application of cryptography to the security of VoIP system in computer networks using LFSR algorithm. The implementation program will be developed Java 2. LFSR algorithm is appropriate for the encryption and decryption of online streaming data, e.g. VoIP (voice chatting over IP). This paper is implemented the encryption module of speech signals to cipher text and decryption module of cipher text to speech signals.
Abstract: The rapid advance of communication technology is
evolving the network environment into the broadband convergence
network. Likewise, the IT services operated in the individual network
are also being quickly converged in the broadband convergence
network environment. VoIP and IPTV are two examples of such new
services. Efforts are being made to develop the video phone service,
which is an advanced form of the voice-oriented VoIP service.
However, the new IT services will be subject to stability and reliability
vulnerabilities if the relevant security issues are not answered during
the convergence of the existing IT services currently being operated in
individual networks within the wider broadband network
environment. To resolve such problems, this paper attempts to analyze
the possible threats and identify the necessary security measures
before the deployment of the new IT services. Furthermore, it
measures the quality of the encryption algorithm application example
to describe the appropriate algorithm in order to present security
technology that will have no negative impact on the quality of the
video phone service.